Efficient encoding method, efficient code decoding method, efficient code encoding apparatus, efficient code decoding apparatus, efficient encoding/decoding system, and recording media

ABSTRACT

Efficient encoding method for carrying out variable bit allocation between channels to samples in the time region or samples in the frequency region of a plurality of channels with respect to information signals of a plurality of channels (CH1˜CH8). This efficient encoding method decomposes bit allocation quantity to channels (CH1, CH3, CH6) in which bit quantity greater than fixed reference quantity (e.g., 147 kbps) is allocated into first bit allocation quantity which is not above 147 kbps and the remaining second bit allocation quantity (bit allocation quantity above 147 kbps) to quantize them. By using this efficient encoding method, it is possible to reproduce, in the state of high sound quality, by making use of inter-channel bit allocation, compressed signals in which sound quality is improved by using inter-channel bit allocation technology with respect to compression of multi-channel system. In addition, also with ordinarily frequently used decoders adapted for carrying out bit allocation within channels by using bit rate less than fixed value individually with respect to respective channels, such compressed signals can be reproduced without great degradation of sound quality.

TECHNICAL FIELD

This invention relates to encoding and decoding of digital speechsignals, and more particularly to a system for carrying out efficientencoding and/or decoding by making use of the hearing sensecharacteristic of the human being with respect to digital speech signalsof a plurality of channels. More specifically, this invention relates toa stereo acoustic system for broadcasting, communication, cinema, videotape recorder or disc player, or a multiple surround acoustic systemcomprised of three channels or more.

Further, this invention relates to an efficient encoding method forreducing bit rate, which is suitable for use in these systems, and anefficient code decoding method and an efficient codedecoding/reproducing method, which correspond to the above-mentionedefficient encoding method.

Furthermore, this invention relates an efficiently encoded signalrecording method for recording signals encoded by such efficientencoding method, and recording media to which recording has beenimplemented.

In addition, this invention relates to an efficiently encoded signaltransmission method for transmitting signals encoded by such efficientencoding method.

BACKGROUND ART

As the efficient encoding method and the efficient encoding apparatusfor musical signals, or audio signals such as speech signals, etc., alarge number of technologies are known.

For example, as one method thereof, there is a blocking frequency banddivision system of dividing an audio signal in the time region intoblocks every predetermined unit time along the time axis to orthogonallytransform signals in the time region every respective blocks intosignals in the frequency region to further divide them into signalcomponents in a plurality of frequency bands to carry outre-quantization/encoding thereof every respective frequency bands. Thissystem is generally called Transform Coding.

This method has been already filed as a patent application by theinventors and the applicant of this application, and is disclosed in,e.g., U.S. Pat. Specification No. 5,301,205.

As another method, there is non-blocking frequency band division methodof dividing an audio signal in the time region into signal components inplural, e.g., about 20 frequency bands without carrying out blockingevery unit time to encode them. In general, method called Sub BandCoding (SBC) is known. For example, such method is disclosed in U.S.Pat. Specification No. 4,896,362 and U.S. Pat. Specification No.5,105,463.

Further, there are also proposed an efficient encoding method and anefficient encoding apparatus in which the Sub Band Coding and theTransform Coding described above are combined. This system is a methodof carrying out band division by the Sub Band Coding thereafter toorthogonally transform signals every respective bands into signals inthe frequency region to implement coding to the orthogonally transformedsignals every respective bands.

As this method, there is a method disclosed in, e.g., U.S. Pat.Specification No. 4,972,484.

Here, as band division filter of the above-described Sub Band Coding,there is a filter, e.g., Quadrature Mirror Filter (QMF), etc. Thisfilter is described in 1976 R. E. Crochiere Digital coding of speech insubbands Bell Syst. Tech. J. Vol. 55, No. 8 1976.

Moreover, in ICASSP 83, BOSTON Polyphase Quadrature filters A newsubband coding technique Joseph H. Rothweiler, method and apparatus forfilter division of equal bands are described.

Moreover, as the above-described orthogonal transform method, there isknown a method of dividing an input audio signal into blocks everypredetermined unit time (frame) to carry out, every blocks, Fast FourierTransform (FFT), Discrete Cosine Transform (DCT), or Modified DiscreteCosine Transform (MDCT), etc. to thereby transform signals on the timebase into signals on the frequency base.

In the embodiment of this application, MDCT is used as orthogonaltransform processing. This MDCT is described in ICASSP 1987Subband/Transform Coding Using Filter Bank Designs Based on Time DomainAliasing Cancellation J. P. Princen A. B. Bradley Univ. of Surrey RoyalMelbourne Inst. of Tech.

Further, as frequency division width in the case of quantizingrespective frequency components, it is effective to determine banddivision width by taking into the hearing sense characteristic of thehuman being. In actual term, bandwidths such that according as frequencyshifts to higher frequency band side, bandwidths become broader, whichare called critical bands, are used. Audio signal of 0˜20 Khz is dividedinto signals in plural (e.g., 25) bands.

Critical bands refer to frequency bands divided in consideration of thehearing sense characteristic of the human being, and are bands thatnarrow band noises of the same intensity in the vicinity of frequency ofa certain pure sound have when the pure sound is masked by those bandnoises.

Moreover, in encoding data every respective bands at this time, codingis carried out by bit quantity determined by predetermined or adaptivebit allocation every respective bands. For example, in encoding MDCTcoefficient data obtained by the MDCT, coding is carried out bydetermined bit quantity,

In regard to the bit allocation, the following two literatures areknown.

In IEEE Transactions of Acoustics, Speech, and Signal Processing, vol.ASSP-15, No. 4, August 1977, technology for carrying out adaptive bitallocation on the basis of magnitudes of signals every respective bandsis described.

Moreover, in ICASSP 1980 the critical band coder--digital encoding ofthe perceptual requirements of the auditory system M. A. Kransner MIT,there is described technology in which hearing sense masking is utilizedto thereby obtain necessary signal-to-noise ratios every respectivebands to carry out fixed bit allocation.

Meanwhile, there is limitation in bit quantity per unit time by bit ratedetermined by recording density in the recording medium, andtransmission path capacity or transmission rate (speed) inbroadcasting/communication. In view of this, in order to satisfy bitrate, efficient encoding method as previously described is used.

However, the above-described conventional bit allocation technology istechnology in which bit allocations every respective channels arerespectively independently taken into consideration (hereinafterreferred to as Independent Allocation for the brevity). In other words,bit quantities of respective channels are certain fixed quantities.Accordingly, such a bit allocation to bridge over a plurality ofchannels is not taken into consideration.

On the contrary, the inventors of this application have alreadyproposed, in the specifications and drawings of Japanese patentapplication Ser. No. 15,492/1993 as the Japanese Patent Application andU.S. patent application Ser. No. 08/184,471 corresponding thatapplication, a method in which common bits which can be used for aplurality of channels are ensured to allocate suitable quantity of bitsto channels for which bits are required.

Since bit allocations are not independent every respective channels,such a method is called inter-channel bit allocation (hereinafterreferred to as Dependent Allocation for the brevity).

The former proposal of the inventors of this application ensures commonbits, thus contributing to improvement in sound quality. However, it canbe also said that the method of ensuring common bits at all times isredundant.

Ordinarily, in the case where there are a plurality of channels,differences occur by information quantities of respective channels inbit quantities that respective channels require.

For example, in the case where audio signal is stereo, it is now assumedthat bit quantity that the right channel requires is 120% of thereference (standard) quantity, and, on the other hand, bit quantity thatthe left channel requires is 50%. In such case, bit quantity allowed forthe right channel is as far as 100% of the reference quantity. As aresult, deterioration in sound quality would take place by 20% ofinsufficient quantity.

On the contrary, there exist excess bit quantity which is as great as50% of the reference quantity, and redundant bit quantity of 50% of thereference quantity is used for that left channel as so called overquality. Of course, redundant bits contribute to improvement in soundquality, but this improvement is nothing but improvement to such adegree which it is not perceived by the hearing sense of the humanbeing.

Meanwhile, in this example, it is conceivable to use bit quantity of 50%of the reference quantity which was redundant at the left channel for20% which is insufficient at the right channel (hereinafter referred toas Subsidiary Allocation for the brevity).

If such a method can be realized, it is possible to satisfy apredetermined bit rate while maintaining high sound quality at bothchannels.

Particularly, in the case of audio signals of music or cinema, etc.,unlike telephone, there are many instances where one information isconstituted by sets of a plurality of channels. Accordingly, there areinstances where redundant bits might take place at any channel orchannels of a plurality of channels. It is therefore considered thatdependent allocation is effective.

Meanwhile, in this case, another problem takes place. Namely,conventional decoder (efficient decoding apparatus) decodes signals of arecording medium on which encoded signals based on independentallocation are recorded.

For this reason, it is impossible to decode, with conventional decoder(decoding apparatus), signals of a recording medium on which signalsencoded by dependent allocation are recorded.

In addition, if decoding apparatus for decoding signals of a recordingmedium on which encoded signals are recorded by dependent allocationcannot decode signals of recording media already on the market, i.e.,recording media on which signals encoded only by independent allocationare recorded, this is considerably disadvantageous to user.

DISCLOSURE OF THE INVENTION

In view of the above, an object of this invention is to provide atechnology capable of obtaining compressed signals caused to have highsound quality by using dependent allocation.

Another object is to provide an efficient encoding technology in whichthe above-mentioned dependent allocation technology is used to permitreproduction of high sound quality, and even if conventional independentallocation is implemented, reproduction can be made in the state whereno great deterioration of sound quality takes place in decoding.

A further object is to provide an encoding method and a decoding methodfor encoded signals according to the technology of this invention.

A further object is to provide an encoding apparatus, a decodingapparatus for encoded signals, and a system composed of an encodingapparatus and a decoding apparatus according to the technology of thisinvention.

A further object is to provide recording media adapted so that encodedsignals formed by the encoding method and the encoding apparatusaccording to the technology of this invention are recorded therein.

A further object is to provide a transmission method and a transmissionapparatus for transmitting encoded signals formed by the encoding methodand the encoding apparatus according to the technology of thisinvention.

An efficient encoding method according to this invention which has beenproposed in order to attain the above-described objects is directed toan efficient encoding method in which, with respect to signals of aplurality of channels, between channels to sample data in the timeregion or sample data in the frequency region of the plurality ofchannels, adaptive bit allocation is carried out. Namely, bit allocationto channel or channels which require(s) bit quantity greater than afixed reference value is decomposed into a first bit allocation quantitywhich is not above the fixed reference value at most, and the remainingsecond bit allocation quantity.

The first bit allocation quantity is caused to be in the range of thefixed reference value corresponding to bit quantity which could used fordata at the time of conventional independent allocation so thatcompatibility with the conventional system is taken into consideration.

The second bit allocation quantity is caused to be the portion above bitquantity which could be used for data at the time of conventionalindependent allocation so that sound quality of corresponding channel istake into consideration.

An efficient encoding method of this invention comprising the stepsdescribed below.

Namely, in a certain sync block (unit block), total bit allocationquantity of all channels is caused to be substantially fixed.

From scale factors and word lengths for sample data relating to thefirst bit allocation quantity, scale factors for sample data relating tothe second bit allocation quantity are determined.

The first bit allocation quantity is caused to be a quantity in whichbit quantity for sub information is taken into consideration.

The second bit allocation quantity is a quantity obtained by subtractingthe first bit allocation quantity from bit allocation quantity thatcorresponding channel requires.

To sample data within small blocks subdivided with respect to the timebase and the frequency base, the same quantization is implemented withinthe small block. In order to obtain sample data within the small block,an approach is employed to carry out analysis of non-blocking frequencycharacteristic thereafter to further carry out analysis of blockingfrequency characteristic with respect to output of analysis of thenon-blocking frequency characteristic.

Frequency bandwidths of analysis of the non-blocking frequencycharacteristic are the same in at least two bands of the lowestfrequency band. Analysis of the non-blocking frequency characteristic isPQF (Polyphase Quadrature Filter). Frequency bandwidths of thenon-blocking frequency characteristic are such that frequency bandwidthin higher frequency band is caused to be broader than that in lowerfrequency band. It is to be noted that QMF (Quadrature Mirror Filter)may be also used in analysis of the non-blocking frequencycharacteristic.

Analysis of the blocking frequency characteristic is MDCT. In analysisof the blocking frequency characteristic, block size is adaptivelyaltered (changed) by time characteristic of input signal. Alterations ofthe block size are independently carried out every outputs of analysisof the at least two non-blocking frequency characteristics.

Sum of the first bit allocation portion and the second bit allocationportion of each channel changes by maximum value of scale factors orsample data of each channel.

Dependent allocation changes by changes in point of time of amplitudesuch as energy values, peak values or mean values, etc. of signals ofrespective channels. Alternatively, such dependent allocation changes bychanges in point of time of scale factors of respective channels.

Bit quantity which can be used for subsidiary allocation is total bitquantity of excess bits of other channels even at the maximum.

An efficient code decoding method of this invention is directed to anefficient code decoding method for decoding encoded signals in whichadaptive dependent allocation to sample data in the time and frequencyregions of a plurality of channels has been implemented to signals of aplurality of channels. This decoding method is characterized in that, atthe time of encoding, bit allocation quantity to channel or channels towhich bit quantity greater than a fixed reference quantity is allocatedis decomposed into a first bit allocation quantity which is not abovethe fixed reference quantity at most and the remaining second bitallocation quantity.

Here, efficient code decoding method of this invention may be featuredbelow.

Total bit quantity with respect to all channels in total of the firstbit allocation quantity and the second bit allocation quantity issubstantially fixed. Scale factors for sample data relating to thesecond bit allocation quantity are determined from scale factors andword lengths for sample data relating to the first bit allocationquantity.

The first bit allocation quantity is bit allocation quantity which doesnot include subsidiary allocation bits, and the second bit allocationquantity is bit allocation quantity including subsidiary allocationbits.

Sample data in which the same quantization has been carried out withinsmall blocks subdivided with respect to the time base and the frequencybase are decoded. Sample data within the small block are caused toundergo blocking frequency synthesis, and output of the blockingfrequency synthesis is caused to be input of non-blocking frequencysynthesis, thus to obtain output of non-blocking frequency synthesis.Frequency bandwidths of the non-blocking frequency synthesis are thesame at least in two bands of the lowest frequency band.

The non-blocking frequency synthesis is PQF. Frequency bandwidths of thenon-blocking frequency synthesis are set so that frequency bandwidth inhigher frequency band is broader than that in lower frequency band. Thenon-blocking frequency synthesis may be QMF. The blocking frequencysynthesis is inverse MDCT. In the blocking frequency synthesis, itsblock size is adaptively altered (changed) by time characteristic ofinput signal. Alterations of the block size are independently carriedout every input bands of the at least two non-blocking frequencysyntheses.

Sum of the first bit allocation quantity and the second bit allocationquantity of each channel is substantially determined by maximum value ofscale factors or sample data of each channel. In the case where aplurality of channels are provided, detection of channel or channels towhich bit quantity greater than a fixed reference quantity is allocatedis carried out by detecting that allocation bit quantity to the channelis greater than or equal to second reference quantity smaller than thefixed reference quantity.

Moreover, in efficient code decoding/reproducing method of thisinvention, there are at least two sample block groups separatelyrecorded within one sync block (continuous signal is caused to undergoblocking every predetermined time unit) and taken out therefrom. Namely,there are first bit allocation sample group for allocating bit quantitygreater than a fixed reference quantity for a plurality of channels, andthe remaining second bit allocation sample group of the first bitallocation sample group for a plurality of channels.

Here, in respective channels, decode/reproduction is carried out fromthe first bit allocation sample group for allocating bit quantitygreater than a fixed reference quantity of each channel and theremaining second bit allocation sample group of the first bit allocationsample group.

An efficiently encoded signal recording method of this invention ischaracterized in that first bit allocation sample group in which bitquantity greater than a fixed reference quantity for a plurality ofchannels is allocated and the remaining second bit allocation samplegroup of the first bit allocation sample group for a plurality ofchannels are recorded in a separate manner within one sync block.

Further, the first bit allocation sample groups and the second bitallocation sample groups are alternately recorded every respectivechannels.

Recording media of this invention is adapted so that signals encoded bythe efficient encoding method of this invention are recorded thereon ortherein.

Such recording media may be cinema film, disc, tape and card includingsemiconductor memory therein.

Namely, in this invention, bit allocation in which subsidiary allocationbits for dependent allocation are included and bit allocation in whichno subsidiary allocation bit is included are determined. Bit allocationsin which no subsidiary allocation bit is included are independentlydetermined every respective channels, and have fixed bit allocationquantities substantially every channels.

With respect to channel or channels in which bit allocation quantity inwhich the subsidiary allocation bits are included is greater than bitallocation quantity in which no subsidiary allocation bit is included,subsidiarily allocated information samples in the time region or in thefrequency region are divided (grouped) into information samples (A)based on bit allocation in which no subsidiary allocation bit isincluded and the remaining information samples (B).

The remaining information sample (B) is determined as information samplehaving magnitude of difference between information sample (C) based onbit allocation in which subsidiary allocation bits are included andinformation sample (A) based on bit allocation in which no subsidiaryallocation bit is included.

On the other hand, with respect to channel or channels in which bitallocation quantity including subsidiary allocation bits is the same asbit allocation quantity including no subsidiary allocation bit or issmaller than that, subsidiarily allocated sample information (C) in thetime region or in the frequency region is used as bit allocation ofcorresponding channel.

From facts as described above, in the case where there is used decoderfor decoding encoded signals in which bit allocation has been carriedout only by independent allocation, this decoder reproduces informationsample (A) based on bit allocation in which no subsidiary allocation bitis included with respect to channel or channels where bit allocationquantity in which subsidiary allocation bits are included is greaterthan bit allocation in which no subsidiary allocation bit is included.

In contrast, with respect to channel or channels where bit allocationquantity in which subsidiary allocation bits are included is equal tobit allocation quantity in which no subsidiary allocation bit isincluded, it is sufficient to reproduce subsidiarily allocatedinformation sample (C).

Moreover, in the case where complete reproduction is carried out, withrespect to channel or channels where bit allocation quantity in whichsubsidiary allocation bits are included is greater than bit allocationquantity in which no subsidiary allocation bit is included, subsidiarilyallocated sample information can be reproduced as reproduced sound ofhigher sound quantity by using both information sample (A) based on bitallocation in which no subsidiary allocation bit is included and theremaining information sample (B). To realize this, it is sufficient toadd information obtained by respectively adding information sample (A)and information sample (B).

Further, total bit allocation quantity with respect to all channels intotal of bit allocation quantity of the information sample (A) and bitallocation quantity of the information sample (B) is caused to besubstantially fixed, thereby making it possible to carry out recordingonto recording media for which it is required that bit rate is fixed.

In the above-mentioned case, an approach is employed to determine scalefactor for normalization of sample data relating to bit allocation ofthe information sample (B) from scale factor and word length for sampledata of the information sample (A), thereby making it possible togenerate, on the decode side, scale factor relating to bit allocation ofthe information sample (B) without sending it from the encode side tothe decode side. Accordingly, information quantity necessary forrecording or transmission can be reduced.

Moreover, in order to obtain sample information (A) based on bitallocation in which no subsidiary allocation bit is included, it iseffective for reduction of quantizing noise to carry out quantizationincluding round-off processing.

Further, in order that the decoder side recognizes channel in which bitallocation for the information sample (B) has been carried out, judgment(recognition) carried out on the basis of the fact that bit allocationquantity to channel is greater than a second reference quantity which issmaller than the fixed reference quantity is profitable in that there isno necessity of sending dedicated data from the encode side to thedecode side.

In addition, in this invention, there is employed an approach in which,to samples within small blocks sub-divided with respect to the time baseand the frequency base, the same quantization is implemented within thesmall block. In order to obtain samples within the small block, anapproach is employed to carry out analysis of non-blocking frequencycharacteristic such as filter, etc. thereafter to conduct analysis ofblocking frequency characteristic such as orthogonal transform, etc.with respect to output of analysis of non-blocking frequencycharacteristic such as the filter, etc.

At this time, the fact that frequency bandwidths of analysis of thenon-blocking frequency characteristic are the same at least in two bandsof the lowest frequency band is advantageous to reduction of cost.Moreover, the fact that frequency bandwidths of analysis of thenon-blocking frequency characteristic are such that frequency bandwidthin higher frequency band is broader than that in lower frequency band isimportant in utilization of the effect of the hearing sense based oncritical bands.

With respect to analysis of the blocking frequency characteristic, byadaptively altering (changing) corresponding block size in dependencyupon time characteristic of input signal, optimum processingcorresponding to time characteristic of input signal can be carried out.Employment of method of independently carry out alterations of the blocksize every output bands of analysis of the at least two non-blockingfrequency characteristics is effective for preventing interferencebetween frequency components to carry out optimum processingindependently every respective band components.

Further, employment of method of determining bit allocation quantitiesgiven to respective channels by maximum value of scale factors or sampledata of respective channels is effective for reduction of operation.

In addition thereto, employment of method of changing bit allocationquantities given to respective channels by change in point of time ofamplitude information represented by scale factors of respectivechannels is also advantageous for the purpose of allowing quantizingnoise to be difficult to detect. Further, the first bit allocationsample group for a plurality of channels and the second bit allocationsample group for a plurality of channels are recorded in a separatemanner into one sync block including information for sync.

In accordance with this invention, an approach is employed to decomposebit allocation quantity to channels for allocating bit quantity greaterthan a fixed reference quantity with respect to signals of a pluralityof channels into first bit allocation quantity which is not above thefixed reference quantity at most and the remaining second bit allocationquantity to carry out variable bit allocation between channels to sampledata in the time region and sample data in the frequency region of aplurality of channels, thereby permitting high sound qualityreproduction utilizing dependent allocation. Further, also in ordinarilyfrequently used decoders in which there is applied adaptive bitallocation technology in the frequency region and in the time regionevery channels by using bit rate less than fixed value individually withrespect to respective channels, reproduction can be carried out withoutgreat deterioration of sound quality. Accordingly, compatibility iskept, thus permitting transmission and reception of information betweendifferent recording media.

Particularly, in motion picture projector, it becomes possible to usedecoder using technology for carrying out adaptive bit allocation in thefrequency and time regions every respective channels by using bit rateless than fixed value every respective channels. Accordingly, it ispossible to provide audio system for which high sound quality isrequired or system suitable for sound recording/reproduction of cinema.

At this time, an approach is employed to determine scale factors forsample data relating to the second bit allocation quantity from scalefactors and word lengths for sample data relating to the first bitallocation quantity to thereby prepare, on the decode side, scalefactors for sample data relating to the second bit allocation withoutsending them from the encode side to the decode side, thus permittingreduction of information quantity necessary for recording ortransmission.

Further, from the fact that bit quantity of first bit allocationquantity which is bit allocation in which no subsidiary allocation bitis included which is not above the fixed reference quantity at most isgreater than second reference quantity which is smaller than the fixedreference quantity, the fact that the decode side recognizes channel inwhich the second bit allocation has been carried out makes itunnecessary to send dedicated data from encode side to decode side.

Implementation of quantization including round-off processing in orderto obtain sample information based on bit allocation in which nosubsidiary allocation bit is included is effective for the purpose ofreducing quantizing noise in decode using adaptive bit allocationtechnology in the frequency and time regions every channels by using bitlate less than fixed value every respective channels.

Further, in order to obtain samples within small blocks subdivided withrespect to the time base and the frequency base, after analysis ofnon-blocking frequency characteristic such as filter, etc. is carriedout, output of analysis of non-blocking frequency characteristic such asthe filter, etc. is caused to undergo analysis of blocking frequencycharacteristic by orthogonal transform, etc., thereby permittingdetermination of quantizing noise in which hearing sense masking in thetime and frequency regions is exhibited. Thus, analysis of frequencycharacteristic preferable from viewpoint of hearing sense can be made.

At this time, the fact that frequency bandwidths of analysis of thenon-blocking frequency characteristic are the same at least in two bandsof the lowest band is useful for reduction of cost.

Frequency bandwidths of analysis of the non-blocking frequencycharacteristic are such that according as frequency shifts to higherfrequency band side, frequency band is caused to be broader at least inthe highest frequency band, thereby making it possible to efficientlyutilize the effect of hearing sense based on critical bands.

In analysis of the blocking frequency characteristic, there is employedan approach such that corresponding block size is adaptively altered(changed) by time characteristic of input, thereby making it possible tocarry out optimum processing corresponding to time characteristic.

Employment of method of independently carrying out alterations of theblock size every output bands of analysis of the at least twonon-blocking frequency characteristics is effective for the purpose ofpreventing interference between frequency components to carry outoptimum processing independently every respective band components.

In calculation of subsidiary allocation bits, those bits are calculatedby scale factors of respective channels, thereby making it possible tosimplify the bit allocation calculation.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block circuit diagram showing an example of theconfiguration of an efficient encoding apparatus according to thisinvention.

FIGS. 2A and 2B are views showing and frequency and time divisions ofsignal in the efficient encoding apparatus according to this invention.

FIG. 3 is a block circuit diagram showing an example of theconfiguration for determining bit allocation parameters at multi-channelof the efficient encoding apparatus according to this invention.

FIGS. 4A and 4B are views showing concept for carrying out bitallocation from magnitude of spectrum components between channels.

FIGS. 5A to 5C are views showing how to determine parameters for bitallocation in which time characteristic of signals between channels istaken into consideration.

FIG. 6 is a view showing the relationship between bit allocationquantity of bit allocation (1) and tonality.

FIG. 7 is a view showing the relationship between bit allocationquantity of bit allocation (1) and rate of time change.

FIG. 8 is a view showing noise spectrum at the time of uniformallocation.

FIG. 9 is a view showing an example of noise spectrum by bit allocationfor obtaining auditory sense effect caused to have dependency withrespect to frequency spectrum and level of signal.

FIG. 10 is a block circuit diagram showing the configuration forrealizing bit allocation method using both magnitude of signal andauditory sense allowed noise spectrum.

FIG. 11 is a block circuit diagram showing the configuration fordetermining allowed noise level.

FIG. 12 is a view showing an example of masking threshold by signallevels of respective bands.

FIG. 13 is a view showing information spectrum, masking threshold andminimum audible limit.

FIG. 14 is a view showing signal level dependent and auditory senseallowed noise level dependent bit allocations with respect to signal oflower tonality.

FIG. 15 is a view showing signal level dependent and auditory senseallowed noise level dependent bit allocations with respect to signal ofhigher tonality.

FIG. 16 is a view showing quantizing noise level with respect to signalof lower tonality.

FIG. 17 is a view showing quantizing noise level with respect to signalof higher tonality.

FIGS. 18A and 18B are views showing the relationship of bit allocationin multi-channel.

FIG. 19 is a block circuit diagram showing the relationship betweenfirst and second quantizing circuits.

FIG. 20 is a view showing, in a model form, arrangement of data to syncblock.

FIG. 21 is a block circuit diagram showing an example of theconfiguration of an efficient code decoding apparatus according to thisinvention.

FIG. 22 is a flowchart of bit allocation of an efficient encoding methodaccording to this invention.

BEST MODE FOR CARRYING OUT THE INVENTION

Embodiments of an efficient encoding apparatus (encoder) to which anefficient encoding method of this invention is applied and a decodingapparatus (decoder) to which an efficient code decoding method(efficient code decoding/reproducing method) of this invention isapplied will now be described with reference to the attached drawings.

In this embodiment, input digital signal such as audio PCM signal, etc.is caused to undergo efficient encoding by using respective technologiesof Sub Band Coding (SBC), Adaptive Transform Coding (ATC) and AdaptiveBit Allocation Coding (APC-AB). This encoding technology will now bedescribed with reference to FIG. 1.

FIG. 1 shows the efficient encoding apparatus of the embodiment to whichthis invention is applied.

In summary, this efficient encoding apparatus is adapted to divide aninput digital signal in the time region into signals in a plurality offrequency bands by QMFs to carry out orthogonal transform processingevery respective frequency bands to allow them to be spectrum data inthe frequency region to adaptively carry out bit allocation of spectrumdata thus obtained every critical bands to encode them.

At this time, in higher frequency bands, bands obtained by furtherdividing critical bands are used. Of course, frequency division width ofnon-blocking by QMF may be equal division width.

Further, in the embodiment of this invention, block size (block length)is adaptively changed in accordance with input signal prior toorthogonal transform processing, and floating processing is carried outin critical band units, or every bands obtained by further subdividingcritical bands in higher frequency bands.

Floating processing is processing for normalizing a plurality of datavalues on the basis of index of 1.

Further, the efficient encoding apparatus of FIG. 1 will be described indetail.

Input terminal 10 is supplied with audio PCM signal of, e.g., 0˜22 Khz.While it is sufficient that ordinary audio frequency band is 0˜20 Khz,higher frequency band side is expanded up to 22 Khz so that higherquality audio signals can be dealt.

This input signal is first divided into signal in 0˜11 Khz band andsignal in 11 k˜22 Khz by band division filter 11. Further, the signal in0˜11 Khz band is divided into signal in 0˜5.5 Khz band and signal in 5.5k˜11 Khz band by band division filter 12 similarly constituted with QMF.

Signals of respective bands from band division filters 11, 12 are sentto MDCT circuits 13˜15 which are orthogonal transform circuit, at whichthey are respectively transformed into MDCT circuits 13 to 15. In thiscase, the above-mentioned signals are caused to undergo MDCT processingon the basis of block sizes determined by block determining circuits19˜21 every respective bands.

Actual examples of block sizes at respective MDCT circuits 13˜15determined by the block determining circuits 19˜21 are shown in FIGS. 2Aand 2B. The case where the orthogonal transform block size is long interms of time base (hereinafter referred to as long mode) is shown inFIG. 2A, and the case where the orthogonal transform block size is shortin terms of time base (hereinafter referred to as short mode) is shownin FIG. 2B.

In the actual example of FIG. 2, the above-mentioned three filteroutputs respectively have two orthogonal transform block sizes.

Namely, in the case of long block length (11.6 msec) as shown in FIG.2A, the number of samples within one block is caused to be 128 sampleswith respect to signal in the 0˜5.5 Khz band of the lower frequency sideand signal in the 5.5 k˜11 Khz band of the medium frequency band. On thecontrary, in the case of short block length (2.9 msec) as shown in FIG.2B, the number of samples within one block is caused to be 32.

Further, with respect to signal in the 11 k˜22 Khz band of the higherfrequency band side, the number of samples within one block is caused tobe 256 samples in the case of long block length (FIG. 2A), and thenumber of samples within one block is caused to be 32 samples in thecase of short block length (1.45 msec) (FIG. 2B).

In the case where short length block is selected in this way, thenumbers of samples of orthogonal transform blocks of respective bandsare caused to be equal to each other so that according as frequencyshifts to higher frequency band, resolution is improved to more degree,and the kind of windows for MDCT processing is decreased.

In this embodiment, signals indicating block sizes determined at theblock determining circuits 19˜21 are delivered to respective MDCTcircuits 13˜15 so that their windows are switched. In addition, thosesignals are sent to adaptive bit allocation encoding circuits 16˜18which will be described later, and are also outputted from outputterminals 25˜27 so that they are used for recording/transmission.

In FIG. 1, for a second time, MDCT coefficient data which are spectrumdata in the frequency region obtained after undergone MDCT processing atrespective MDCT circuits 13˜15 are combined every critical bands orevery bands obtained by further dividing critical bands in higherfrequency bands, and the data thus combined are sent to adaptive bitallocation encoding circuits 16˜18.

The adaptive bit allocation encoding circuits 16˜18 normalize andre-quantize respective MDCT coefficient data in dependency uponallocated bit quantities every critical bands or every bands obtained byfurther dividing critical bands in higher frequency bands.

At this time, adaptive bit allocation encoding circuits 16˜18 carry outallocation of bit quantities every respective blocks with bit quantitiesdesignated to the respective channels being as respective upper limitvalues.

Digital signals indicating spectrum distribution (MDCT coefficients) ofrespective channels are delivered to adaptive bit allocating circuit 30through terminal 29. On the other hand, from the adaptive bit allocatingcircuit 30, bit quantities which can be used in blocks of respectivechannels are delivered to adaptive bit allocation encoding circuits16˜18 through terminal 28.

Data encoded by predetermined bit quantities in this way are taken outthrough output terminals 22˜24. At the same time, from the adaptive bitallocation encoding circuits 16˜18, scale factor signals relating tonormalization and word length signals indicating word length inimplementation of re-quantization are obtained. These signals are alsoas sub information from output terminals 22˜24.

Moreover, outputs of respective MDCT circuits 13˜15 in FIG. 1 can beobtained by a method of determining energies every critical bands asmentioned above or every bands obtained by further dividing criticalbands by calculating root mean square values of respective amplitudevalues every these bands, or similar method.

Of course, in place of the energy, the above-mentioned scale factorsthemselves may be used for bit allocation at times subsequent thereto.In this case, since operation of energy calculation becomes unnecessary,hardware scale can be cut down. In addition, peak values or mean valuesof amplitude values may be used in place of energies every respectivebands.

Actual configuration and operation of adaptive bit allocating circuit 30for carrying out the above-mentioned bit allocation will now bedescribed with reference to FIG. 3. In the example of FIG. 3, cinema istaken as an example and the number of channels of audio signal is causedto be 8 (CH1˜CH8).

In FIG. 3, explanation will now be given by using channel CH1 withrespect to the portion common to respective channels (the same referencenumerals are respectively attached to other channels and theirexplanation is omitted).

Input signals from respective channels are delivered to input terminals31 of corresponding respective channels. It should be noted that theseterminals 31 correspond to terminal 29 of FIG. 1. Such input signal isinputted to mapping circuit 32. Thus, frequency analyses (spectrumdistributions) of input signal are obtained.

Here, in the case where filter is used as the mapping circuit, timeregion sample data is obtained as subband signal. Moreover, in the casewhere orthogonal transform is used, and in the case where orthogonaltransform is used after filtering, frequency region sample data isobtained.

These sample data are combined every plural sample data by blockingcircuit 33. As previously described, in the case where filter is used,plural sample data in the time region are combined. In the case whereorthogonal transform is used and orthogonal transform after filtering isused, plural samples in the frequency region are combined.

Moreover, time changes (V) of sample data caused to sequentially undergomapping in accordance with input signal are calculated by time changecalculating circuit 34. By reflecting transient change of input signalin bit allocation, higher quality signal can be obtained.

Respective samples combined every plural samples by the blocking circuit33 are normalized at normalizing circuit 37. In this example, scalefactors (SF) which are coefficients for normalization are obtained byscale factor calculating circuit 35. Scale factors of 1 common to pluralsamples are used, thus to efficiently compress digital signal.

At the same time, tonality is calculated at tonality calculating circuit36. Tonality indicates undulation (ups and downs) of spectrumdistribution of input signal. Input signal having great undulation iscalled signal having high tonality. Its detail will be described later.

Parameters such as time change (V) scale factor (SF) and tonality (T),etc. of sample data determined in a manner as described above are usedfor calculation of bit allocation at bit allocation circuit 38.

Bit allocation calculation is basically adaptive bit allocationcorresponding to input signal. In more detail, there are independentallocation corresponding to spectrum distributions or transientcharacteristics of input signals of respective channels, and dependentallocation utilizing correlation between respective channels. Further,any adjustment is implemented to allocation by degree of importance ofrespective channels, intended purpose and bandwidth of signal, etc.

When it is now assumed that MDCT coefficients are represented so thatbit quantity which can be used for transmission or recording is 800 kbpsin all channels, bit allocation circuit 38 of this embodiment determinestwo bit allocations of first bit allocation (first bit allocationquantity) including dependent allocation bits and second bit allocation(second bit allocation quantity) including no dependent allocation bit.

These bit allocation quantities are delivered to adaptive bit allocationencoding circuits through terminal 39 (terminal 28 in FIG. 1) everyrespective channels.

The first bit allocation including dependent allocation bits will beinitially described. Here, bit allocation is adaptively carried out bymaking reference to distribution in the frequency region of scalefactors (SF).

In this case, dependent allocation is carried out in accordance withdistributions in the frequency region of scale factors (SF) of allchannels, thereby making it possible to carry out effective bitallocation.

At this time, let now consider the case where signal information of aplurality of channels are mixed within the same sound filed, and arereached to the left and right ears as in the case where thoseinformation are reproduced by speaker. In this case, it is consideredthat masking effect is exerted by added result of all channel signals.

Accordingly, it is effective that bit allocation is carried out so thatrespective channels have the same noise level within the same band asshown in FIGS. 4A and 4B.

This is because in the case where noise level of a certain channel isgreater than those of other channels, signal would be perceived as noiseat that channel, and even if noise level of a certain channel is causedto be smaller than those of other channels, the entire noise level wouldbe eventually determined by noise levels of other channels.

As one method therefor, it is sufficient to carry out bit allocationproportional to the magnitude of scale factor index. Namely, bitallocation is carried out by the following formula.

    Bm=B*(ΣSfn)/St

    St=Σ(ΣSfn)

In the above-mentioned formula, Bm is bit allocation quantities torespective channels, B is bit allocation quantity to all the channels,and Sfn is scale factor index, which substantially corresponds logarithmof peak value. n is block bloating band numbers every respectivechannels, m is channel number, and St is sum of scale factor indices ofall channels. It is to be noted that, in FIGS. 4A and 4B, only channelCH1 and channel CH8 are illustrated, and illustration of channelsCH2˜CH7 is omitted.

In addition to the above, bit allocating circuit 38 has a process todetect time change characteristics (V) of signals of respective channelsto vary independent allocation quantities by their indices. A method ofdetermining index indicating time change will now be described withreference to FIGS. 5A to 5C.

When it is now assumed that there are 8 channels as shown in FIGS. 5A to5C, bit allocation time block which is time unit of bit allocation isdivided into four sections in point of time with respect to informationinput signals of respective channels to obtain peak values of respectivesmall time blocks (sub blocks).

Then, bits are allocated between channels in dependency upon themagnitudes of differences where peak values of respective sub blockschange from small value to great value. When it is now assumed that whenC bits in total of 8 channels can be used for such bit allocation, andmagnitudes of differences where peak values of respective sub blocks ofrespective channels change from small value to great value arerespectively designated at a(FIG. 5A), b(FIG. 5B), . . . h(FIG. 5C)decibel (Db), respective bits can be allocated in a manner as indicatedby the following expression:

C*a/T, C*b/T . . . , C*h/T bits

(T=a+b+c+d+e+f+g+h in the above expression).

According as change of peak value becomes greater, bit allocationquantity with respect to corresponding channel becomes greater. It is tobe noted that, in FIGS. 5A to 5H, only FIGS. 5A, 5B, 5C corresponding tochannels CH1, CH2, CH8 are shown and figures corresponding to channelsCH3˜CH7 are not shown.

The second bit allocation method including no dependent allocation bitwill now be described. As the second bit allocation method, two bitallocation methods will be further described.

These two bit allocations are respectively assumed to be bit allocation(2-1) and bit allocation (2-2). In the following bit allocation, bitquantities allocated to respective channels are caused to be valuesfixed at respective channels.

For example, relatively great bits of 147 kbps are allocated twochannels which take the important portion such as sound, etc. of 8channels, and bit of 2 kbps at most are allocated to subwoofer channel,and bits of 100 kbps are allocated to channels except for the above.

Initially, bit quantity to be used for bit allocation (2-1) isestablished. To realize this, time tonality information (T) and timechange information (V) of signal information (b) of spectrum informationof signal information (a) are used.

Tonality information (T) will now be described. Value obtained bydividing sum of absolute values of differences between adjacent valuesof signal spectrum (components) by the number of signal spectrumcomponents is used as index. As a simpler method, there is used meanvalue of differences between adjacent scale factor indices of scalefactors every blocks for block floating. Scale factor indexsubstantially corresponds to logarithmic value of scale factor.

In this embodiment, bit quantities to be used for bit allocation (2-1)are respectively set to 80 kbps at the maximum and 10 kbps at theminimum in a manner caused to correspond to value indicating tonality.Here, for simplicity, respective allocation quantities of all channelsare equally set to 100 kbps.

Tonality calculation is carried out in a manner as indicated byfollowing formula:

    T=(1/Wlmax)(ΣABS(Sfn-1)

In the above-mentioned formula, Wlmax is word length maximum value=16,and Sfn is scale factor index, which substantially corresponds tologarithm of peak value. n is block floating band number.

Tonality information (T) and bit allocation quantity of bit allocation(2-1) determined in this way are caused to correspond to each other asshown in FIG. 6.

Along with the above, in this embodiment, divisional ratio between bitallocation (2-1) and any other at least one bit allocation added theretois dependent upon time change characteristic (V) of signal. In thisembodiment, every time intervals obtained by further dividing orthogonaltransform time block size, peak values of signal information arecompared with each other every adjacent blocks. From facts as above,two-dimensional comparison between the time base and the frequency baseis carried out to detect the portion where amplitude of signal suddenlybecomes great to determine divisional ratio by that state.

Time change rate calculation is carried out in a manner as indicated bythe following formula.

    Tt=ΣVm

    Vav=(1/Vmax)*(1/Ch)Vt

In the above formula, Vt is sum relating to channels of changes fromsmall value to great value of peak values of time sub blocks ofrespective channels indicated in terms of Db value, Vm is magnitude ofthe maximum change of changes from small value to great value of peakvalues of time sub blocks of respective channels indicated in term of Dbvalue (the maximum value is limited to 30 Db, and is indicated by Vmax).m is channel number, Ch is the number of channels, Vav is channel meanvalue of changes from small value to great value of peak values of timesubblock indicated in terms of Db value.

The time change rate Vav and the bit allocation quantity of bitallocation (2-1) determined in this way are caused to correspond to eachother as shown in FIG. 7. Allocation to the bit allocation (2-1) isultimately determined by the following formula:

    B=1/2(Bf+Bt)

In the above formula, B is the ultimate allocation quantity to bitallocation (2-1), Bf is bit allocation quantity determined by Tva, andBt is bit allocation quantity determined by Vav.

With respect to the bit allocation (2-1) in this case, allocation in thefrequency and time regions dependent upon scale factors is carried out.

If bit quantity used for bit allocation (2-1) is determined in this way,allocation with respect to bits which were not used in the bitallocation (2-1), i.e., bit allocation (2-2) is determined. In the bitallocation (2-2), various bit allocations as described below are carriedout.

First, portions of bits which can be used in bit allocation (2-2) areused to carry out uniform allocation with respect to all sample values.

An example of quantizing noise spectrum in this case is shown in FIG. 8.In this case, noise level is uniformly reduced over the entire frequencyband. Namely, noise level (dotted lines NL1) at the first bit allocationis uniformly reduced to noise level indicated by dotted lines NL2 byuniform allocation.

Such a uniform allocation is effective when input signal is a lowtonality and smooth signal.

Secondly, remaining bits which can be used in bit allocation (2-2) areused to carry out bit allocation for obtaining auditory sense effectcaused to have dependency with respect to frequency spectrum and levelof signal information.

An example of quantizing noise spectrum with respect to bit allocationin this case is shown in FIG. 9. In this example, bit allocation causedto be dependent upon spectrum of signal is carried out. Bit allocationis carried out particularly placing emphasis upon the lower frequencyside of spectrum of signal to compensate decrease in masking effect atthe lower frequency band side occurring as compared to the lowerfrequency band side. This is based on non-symmetric property of maskingcurve in which importance is attached to the lower frequency band sideof spectrum.

As stated above, in the example of FIG. 9, bit allocation in whichimportance is attached to the lower frequency band is carried out.Namely, noise level (dotted lines NL1) at the first bit allocation isreduced to noise level indicated by dotted lines NL2 by bit allocationfor obtaining effect from a view point of the auditory sense caused tohave dependency with respect to frequency spectrum and level.

It is to be noted that thick lines indicated by SS in FIGS. 8 and 9 showconnection of peaks of signal spectrum components.

Ultimately, sum of bit allocation quantity of bit allocation (2-1) andbit allocation quantity added to bit allocation (2-2) is obtained at bitallocation circuit 38 of FIG. 3. Ultimate bit allocation quantity isgiven as sum of respective bit allocation quantities.

Another method of bit allocation including no dependent allocation bitwill now be described.

The operation of adaptive bit allocating circuit 800 in this case willnow be described with reference to FIG. 10.

In the embodiment of FIG. 10, input signal is given as sum of outputs(MDCT coefficients) of MDCT circuits 13˜15. MDCT coefficients aredelivered to input terminal 801. MDCT coefficients delivered to theinput terminal 801 are given to circuit 803 for calculating energiesevery bands. This circuit 803 for calculating energies every bandscalculates signal energies relating to critical bands or respectivebands obtained by further carrying out re-division of critical bands inhigher frequency bands. Energies relating to respective bands calculatedat the circuit 803 for calculating energies every bands are delivered toenergy dependent bit allocation circuit 804.

The energy dependent bit allocating circuit 804 carries out a bitallocation to prepare white quantizing noises by using usable total bitsfrom usable total bit generating circuit 802, a certain percentage of128 kbps, e.g., 100 kbps. At this time, according as tonality of inputsignal becomes higher, i.e., unevenness of spectrum of input signalbecomes greater, ratio occupied in the 128 kbps of this bit quantityincreases.

In order to detect unevenness of spectrum of input signal, sum ofabsolute values of differences between block floating coefficients ofadjacent blocks is used as index. Then, bit allocations proportional tologarithmic values of energies of respective bands are carried out withrespect to the determined usable bit quantity.

Circuit 805, which calculates auditory sense allowed noise leveldependent bit allocation, first determines allowed noise quantitiesevery respective critical bands in which so called masking effect, etc.is taken into consideration on the basis of spectrum data divided everycritical bands. Then, bits in which energy dependent bits are subtractedfrom usable total bits so as to give auditory sense allowed noisespectrum are allocated. The energy dependent bits and the bits dependentupon auditory sense allowed noise level which are determined in this wayare added. The added signal thus obtained is outputted from outputterminal 807 as bit allocation signal.

With respect to bit allocation signal, in the adaptive bit allocationencoding circuits 16˜18 of FIG. 1, in dependency upon bit quantitiesallocated every respective critical bands or every plural bands obtainedby further dividing critical bands in higher frequency bands, respectivespectrum data (or MDCT coefficient data) are re-quantized. Data encodedin this way are taken out through output terminals 22˜24 of FIG. 1.

Explanation will now be given in more detail in connection with auditorysense allowed noise spectrum calculating circuit in the auditory senseallowed noise spectrum dependent bit allocation circuit 805. MDCTcoefficients obtained at MDCT circuits 13˜15 are given to the allowednoise spectrum calculating circuit.

FIG. 11 is a block circuit diagram showing outline of the configurationof an actual example in which the allowed noise calculating circuit isdescribed in a collective manner. In FIG. 11, input terminal 521 issupplied with spectrum data of the frequency region from MDCT circuits13˜15.

These input data of the frequency region are sent to energy calculatingcircuit 522 every band, at which energies every critical bands aredetermined, e.g., by a method of calculating sum total of respectiveamplitude value squares within the corresponding bands, or similarmethod. There are instances where-peak values or mean values, etc. ofamplitude values may be used in place of energies every respectivebands. As output from the energy calculating circuit 522, e.g., spectrumof sum total value of respective bands is generally called barkspectrum. FIG. 12 shows bark such spectrum (components) SB everyrespective critical bands. It should be noted that, in FIG. 12, for thebrevity of illustration, the number of bands of the critical bands isrepresented by 12 (B1˜B12).

In order to allow for influence in so called masking of the barkspectrum SB, such a convolution processing to multiply the bark spectrum(components) SB by weighting function to add multiplied values isimplemented. To realize this, outputs of the energy calculating circuit522 every bands, i.e., respective values of the bark spectrum componentsSB are sent to convolution filter circuit 523. This convolution filtercircuit 523 is composed of, e.g., a plurality of delay elements forsequentially delaying input data, a plurality of multipliers (e.g., 25multipliers corresponding to respective bands) for multiplying outputsfrom these delay elements by filter coefficients (weighting function),and a sum total adder for taking sum total of respective multiplieroutputs.

The above-mentioned masking refers to the phenomenon that a certainsignal is masked by another signal by the characteristic from aviewpoint of the hearing sense of the human being so that it cannot beheard. As this masking effect, there are time axis masking effect byaudio signal in the time region and simultaneous masking effect bysignal in the frequency region. By these masking effects, even if anynoise exists at the portion to be masked, such noise would not be heard.For this reason, in actual audio signal, noises within the range to bemasked are considered to be allowable noise.

An actual example of multiplication coefficients (filter coefficients)of respective multipliers of the convolution filter circuit 523 isgiven. When coefficient of multiplier M corresponding to an arbitraryband is assumed to be 1, outputs of respective delay elements aremultiplied by coefficient 0.15 at multiplier M-1, coefficient 0.0019 atmultiplier M-2, coefficient 0.0000086 at multiplier M-3, coefficient 0.4at multiplier M+1, coefficient 0.06 at multiplier M+2, and coefficient0.007 at multiplier M+3. Thus, convolution processing of the barkspectrum SB is carried out. In this example, M is arbitrary integerswhich take 1 to 25.

Then, output of the convolution filter circuit 523 is sent to subtracter524. This subtracter 524 serves to determine level α corresponding toallowable noise level which will be described later in the convolutedregion. It is to be noted that level α corresponding to the allowablenoise level (allowed noise level) is such a level to become equal toallowed noise level every band of critical bands by carrying out inverseconvolution processing in a manner which will be described later. Inthis example, the subtracter 524 is supplied with allowed function(function representing masking level) for determining the level α. Byincreasing or decreasing the allowed function value, control of thelevel α is carried out. This allowed function is delivered from (n-ai)function generating circuit 525 as described below.

Namely, when numbers given in succession from lower frequency band ofbands of critical bands are assumed to be i, level α corresponding toallowed noise level can be determined by the following formula:

    α=S-(n-ai)

In the above formula, n and a are constants (a>0), S is intensity ofconvolution processed bark spectrum, wherein (n-ai) in the formula isallowed function. As an example, n=38 and a=-0.5 may be used.

The above-mentioned level α is determined in this way and this data issent to divider 526. The divider 526 serves to implement inverseconvolution to the level α in the convoluted region. Accordingly, bycarrying out this inverse convolution processing, masking threshold canbe obtained from the level α. Namely, this masking threshold becomesallowed noise spectrum. It is to be noted that while the above-mentionedinverse convolution processing requires complicated operation,simplified divider 526 is used in this embodiment to carry out inverseconvolution.

Then, the masking threshold is sent to subtracter 528 through synthesiscircuit 527. In this case, this divider 528 is supplied with output fromthe energy detecting circuit 522 every band, i.e., the previouslydescribed bark spectrum SB through delay circuit 529.

Accordingly, subtractive operation between the masking threshold andbark spectrum SB is carried out at the subtracter 528. Thus, as shown inFIG. 13, the portion below the level indicated by the level of themasking threshold MS is masked. In this example, delay circuit 529 isprovided for delaying bark spectrum SB from energy detecting circuit 522by taking into consideration delay quantities at respective circuits ofthe synthesis circuit 527 and stages preceding thereto.

Output from the subtracter 528 is taken out through allowed noisecorrection circuit 530 and through output terminal 531, and is sent toROM, etc. (not shown) in which allocation bit quantity information isstored in advance. This ROM, etc. outputs allocation bit quantityinformation every respective bands in dependency upon outputs obtainedthrough allowed noise correction circuit 530 from the subtractingcircuit 528 (levels of differences between energies of respective bandsand output of noise level setting means (not shown).

Energy dependent bits and bits dependent upon auditory sense allowednoise level are added in this way. Thus, allocation bit quantityinformation thus obtained is sent to the adaptive bit allocationencoding circuits 16˜18 through terminal 28 of FIG. 1, at whichrespective spectrum data in the frequency region from MDCT circuits13˜15 are quantized by bit quantities allocated every respective bits.

Namely, in short, adaptive bit allocation encoding circuits 16˜18quantize spectrum data every respective bands by bit quantitiesallocated in dependency upon levels of differences between energies orpeak values every respective bands of the critical bands (criticalbands) or every plural bands obtained by further dividing critical bandsin higher frequency band.

Meanwhile, in synthesis at the above-described synthesis circuit 527, itis possible to synthesize data indicating so called minimum audiblecurve which is the hearing sense characteristic of the human being asshown in FIG. 13 delivered from minimum audible curve generating circuit532 and the above-mentioned masking threshold MS. In this minimumaudible curve, if noise absolute level is below the minimum audiblecurve, such noise would not be heard.

Even if encoding method is the same, this minimum audible curve changesin dependency upon difference of reproduction sound quantity at the timeof reproduction. However, in a digital audio system such that music iscaused to be sound source in practice, in the case of digital recordingby, e.g., 16 bit quantization, there is no great difference in theminimum audible curve.

Accordingly, it is considered that if quantizing noise in the frequencyband, which is most easily perceived, in the vicinity of 4 Khz cannot beheard, quantizing noise less than the minimum audible curve of otherfrequency bands cannot be heard.

Accordingly, when it is assumed to employ way of use such that, e.g.,noise in the vicinity of 4 Khz of dynamic range that system has is notheard, and both the minimum audible curve RC and the masking thresholdMS are synthesized to obtain allowed noise level, allowed noise levelsin this case can be up to the portion indicated by slanting lines inFIG. 13. In this embodiment, level of 4 Khz of the minimum audible curveis caused to be in correspondence with the minimum level correspondingto, e.g., 20 bits. Additionally, in FIG. 13, signal spectrum SS is showntogether.

Moreover, the allowed noise correction circuit 530 corrects allowednoise level in output from the subtracter 528 on the basis ofinformation of, e.g., equi-loudness curve sent from correctioninformation output circuit 533. Here, equi-loudness curve ischaracteristic curve relating to the hearing sense characteristic of thehuman being, and is obtained by determining sound pressures of sound atrespective frequencies which is heard at the same pitch (loudness) asthat of pure sound of, e.g., 1 Khz to connect them by curves. Thisequi-loudness curve is also called equi-sensitivity curve of loudness.

Moreover, this equi-loudness curve depicts substantially the same curveas the minimum audible curve RC shown in FIG. 13. In this equi-loudnesscurve, even if sound pressure is lowered by 8˜10 Db as compared to thatat 1 Khz, e.g., in the vicinity of 4 Khz, sound is heard at the samepitch (loudness) as that at 1 Khz. In contrast, in the vicinity of 50Hz, if sound pressure is not higher than that at 1 Khz by about 15 Db,sound cannot be heard at the same pitch.

For this reason, it is seen that noise beyond level of the minimumaudible curve (allowed noise level) is caused to have frequencycharacteristic given by curve corresponding to the equi-loudness curve.From facts as described above, it is seen that a method of correctingthe allowed noise level by taking the equi-loudness curve intoconsideration is in conformity with the hearing sense characteristic ofthe human being.

Spectrum shape dependent upon the above-described auditory sense allowednoise level is prepared by bit allocation of a certain ratio withinusable total bit 128 kbps. This ratio is decreased to more degreeaccording as tonality of input signal becomes higher.

Bit quantity division method between two bit allocation methods will nowbe described.

Turning back to FIG. 10, signal from input terminal 801 supplied withMDCT circuit output is also given to spectrum smoothness calculatingcircuit 808, at which smoothness of spectrum is calculated. In thisembodiment, value obtained by dividing sum of absolute values ofdifferences between adjacent values of absolute values of spectrum bysum of absolute values of signal spectrum components is calculated assmoothness of the spectrum.

Output of the spectrum smoothness calculating circuit 808 is given tobit division ratio determining circuit 809, at which bit divisionalratio between energy dependent bit allocation and bit allocation byauditory sense allowed noise spectrum is determined. The bit divisionalratio is determined so that in the case where output value of spectrumsmoothness calculating circuit 808 is great, according as that outputvalue becomes greater, smoothness of spectrum is considered to be gone(lost) to more degree, thus to carry out bit allocation in whichemphasis is placed on bit allocation by auditory sense allowed noisespectrum rather than energy dependent bit allocation. Bit divisionalratio determining circuit 809 sends control output to multipliers 811and 812 for respectively controlling magnitudes of energy dependent bitallocation and bit allocation by auditory sense allowed noise spectrum.In this case, when spectrum is assumed to be smooth, and bit divisionalratio determining circuit 809 to multiplier 811 takes value of 0.8 inorder to place emphasis on energy dependent bit allocation, output ofbit divisional ratio determining circuit 809 to multiplier 812 isassumed to be expressed below:

    1-0.8=0.2

Outputs of these two multipliers are added at adder 806, resulting inultimate bit allocation information. The information thus obtained isoutputted from output terminal 807.

State of bit allocation at this time is shown in FIGS. 14 and 15.Moreover, state of quantizing noise corresponding thereto is shown inFIGS. 16 and 17.

FIG. 14 shows the case where signal spectrum indicates lower tonality,and FIG. 15 shows the case where signal spectrum indicates highertonality. In the figures of FIGS. 14 and 15, QS indicated by slantinglines represent signal level dependent bit quantities.

In the figure, QN designated as void indicates auditory allowed noiselevel dependent bit quantities. In the figures of FIGS. 16 and 17, SSindicates signal level, NS indicates noise lowering quantity by signallevel dependent bit allocation, and NN indicates noise lowering quantityby auditory sense allowed noise level dependent bit allocation.

Initially, in FIG. 14 showing the case where spectrum of signal isrelatively flat, bit allocation dependent upon auditory sense allowednoise level is useful for taking large signal-to-noise ratios over theentire bands. However, relatively smaller bit allocation is used in thelower and higher frequency bands. This is because sensitivity withrespect to noise in these bands is small form a viewpoint of hearingsense. While signal energy level dependent bit allocation is small asquantity, bits are preponderantly allocated in the frequency region ofhigher signal level in the medium and lower frequency bands in this caseso as to produce white noise spectrum.

On the contrary, as shown in FIG. 15, in the case where signal spectrumindicates higher tonality, signal energy level dependent bit allocationquantity becomes great, and lowering of quantizing noise is used for thepurpose of reducing noise in extremely narrow bands. Concentration ofauditory allowed noise level dependent bit allocation is not so severethan that.

As shown in FIG. 10, improvement in the characteristic at isolatedspectrum input signal is attained by sum of both bit allocations.

Two allocations of bit allocation including dependent allocation bitsobtained and bit allocation including no dependent allocation bits whichare obtained in this way are used to carry out first and secondquantizations in a manner as described below.

FIGS. 18A and 18B show the number of bits that respective channelsrequire with respect to audio signals of 8 channels, i.e., necessarywithout lowering sound qualities of respective channels.

In FIG. 18A, only lattice pattern portion (CH2, CH4, CH5, CH7, CH8), orsum of both lattice pattern portion and slanting line pattern portion(CH1, CH3, CH6) indicate bit quantities that respective channelsrequire. Among them, slanting line pattern portion indicates bitquantity corresponding to subsidiary allocation bits. Dotted patternportion indicates excess bits (R). This excess bit quantity isdifference between upper limit bit quantities (147 kbps) of each channelallowed at bit rate and bit quantity that each channel actually requires(upper limit value thereof in the case where allowed bit quantity withineach channel is limited to smaller value, e.g., 2 kbps).

In the example of FIG. 18A, channels which requires bit quantity above147 kbps of all 8 channels are channel CH1, channel CH3 and channel CH6.

Initially, with respect to channels in which bit allocation quantityincluding subsidiary allocation bits is above 147 kbps, which isrequired by input signal, e.g., with respect to channel CH,consideration will be made on the assumption that bit quantity isdivided into two portions of portion (I) where a fixed bit quantity,e.g., 147 kbps is maximum and portion (S) above 147 kbps.

Namely, it is now assumed that input signal is divided into the portionquantized by bit quantity of 147 kbps corresponding to independentallocation and the portion quantized by bit quantity of subsidiaryallocation so that respective portions are quantized by those bitquantities.

Let now consider such a processing to decompose, e.g., input signal ofdigital word of 16 bits into 10 bit portion including MSB and 6 bitportion including LSB to quantize the 10 bit portion by independentallocation and to quantize the 6 bit portion by subsidiary portion.

The configuration for carrying out such processing is shown in FIG. 19.

In the configuration of FIG. 19, with respect to respective samples ofbit allocation in which bit quantity is above 147 kbps, normalizationprocessing with respect to blocks every plural samples, i.e., blockfloating processing are carried out. At this time, as coefficientsindicating to what degree block floating is carried out, scale factorsare obtained.

In FIG. 19, signal delivered to input terminal 900 is gain-controlled atgain controller 905.

Then, quantizer 901 carries out re-quantization by respective sampleword lengths by bit quantity (147 kbps) at bit allocation which includesno subsidiary allocation bit. At this time, re-quantization by round-offis carried out in order to allow quantizing noises to be lesser.

Moreover, several bits of the MSB side is caused to be sample wordlength. Sample word length in this case may be fixed. In this case, itis difficult to allow bit quantity to be value close to the bit quantity(147 kbps).

In view of this, the best way is to allow sample word length to beadaptively variable in dependency upon input signal. To realize this, anapproach is employed to integrate data quantity from, e.g., thequantizer 901 to carry out feedback control of the quantizer 901 so thatit becomes close to 147 kbps.

Then, difference between input and output of quantizer 901 is taken atdifference element 902. The difference thus obtained is gain-controlledat gain controller 906, and is then delivered to second quantizer 903.

At this quantizer 903, sample word length of difference between sampleword length of the input signal and sample word length by bit allocationwhich includes no subsidiary allocation bit is obtained. Namely, severalbits of the LSB side of the input signal are obtained.

Floating coefficients at this time is automatically determined fromfloating coefficient and word length used at quantizer 901. Namely, whenit is assumed that word length used at first quantizer 901 is N bits,floating coefficient used at second quantizer 903 is obtained as (2**N).

The second quantizer 903 carries out re-quantization including round-offprocessing by bit allocation similarly to the first quantizer 901.

At channels where required bit quantity is above 147 kbps by twoquantizations in this way, data is divided into data by bit allocationwhich is less than 147 kbps and is as close as 147 kbps, and data by theremaining bit allocation.

Moreover, channels in which there has been a bit allocation such thatrequired bit quantity is smaller than 147 kbps use that bit allocationas they are.

In the above-mentioned example, bits by the dependent allocation arecaused to be included in both bits corresponding to independentallocation indicated by lattice pattern portion (I) and subsidiaryallocation bits (S) indicated by slanting line pattern portion.

Namely, all bits of respective channels are caused to be sum ofindependent allocation bits and dependent allocation bits irrespectiveof presence or absence of subsidiary bits.

Accordingly, with respect to, e.g., the previously described channelCH1, 70% of lattice pattern portion (I) are caused to be bits byindependent allocation, and the remaining 30% are caused to be bits bydependent allocation. Moreover, with respect to subsidiary bits(slanting line portion (S)), 70% are caused to be bits by independentallocation and the remaining 30% are caused to be bits by dependentallocation. In this case, there may be employed an approach in which 50%of subsidiary bits are caused to be bits by independent allocation andthe remaining 50% are caused to be bits by dependent allocation.

In order to further simplify processing, with respect to the portionless than 147 kbps indicated by lattice pattern portion (I), bits byindependent allocation are assumed to be allocated. Moreover, withrespect to subsidiary allocation bits (S) indicated by slanting linepattern portion, bits by dependent allocation are assumed to beallocated. Namely, with respect to all bits of respective channels,subsidiary bits are assumed to be all dependent allocation bits.

Accordingly, in the previously described example of FIG. 18A, forexample, with respect to only channel CH1, channel CH3 and channel CH6,bit allocation in which correlation between channels is taken intoconsideration, i.e., dependent allocation is used to carry out bitallocation to allow corresponding bits to be subsidiary allocation bits.The merit of this method is that calculation for bit allocation becomeseasy.

In this case, as previously described, with respect to magnitudes ofcomponents of the remaining bit allocation, since scale factor can becalculated from scale factor and word length of bit allocation (1) asshown in FIG. 19, only word length is required for decoder.

An example of flowchart for carrying out the above-mentioned bitallocation.

Initially, required bit quantities are calculated every respectivechannels (S10). Then, channels for which bit quantity above referencequantity is required are specified (designated) (S11). Subsequently, sumtotal (ΣR) of excess bits (R) of respective channels is calculated(S12). In this case, with respect to channels for which bit quantityabove reference quantity is required, sum total (ΣS) of bit quantities(S) above reference quantity is determined (S13).

Sum total of excess bits (R) and sum of bit quantities (S) abovereference quantity are compared with each other (S14). If sum of bitquantities (S) above reference quantity is greater than sum total ofexcess bits (R), bit quantities (S) above reference quantity ofrespective channels are reduced until sum total of bit quantities (S)above reference quantity is less than sum total of excess bits (R)(S15).

When bit allocation quantities to respective channels are determined,independent allocation is first carried out by a portion of allocationbit quantity (S16). Subsequently, dependent allocation is carried out bythe remainder (S17).

When allocations to respective channels are completed, judgments as towhether subsidiary allocation is carried out are carried out everyrespective channels (S18). With respect to channels in which bitallocation above reference quantity has been carried out, subsidiaryallocation is carried out (S19). With respect to channels in which bitallocation which is not above reference quantity has been carried out,no subsidiary allocation is carried out (S20).

In the example of FIG. 18A, reference is not made to sub information.However, in practice, not only bits for data but also bits for subinformation for restoring (reconstructing) that data must be taken intoconsideration.

In view of this, the example in which sub information is taken intoconsideration is shown in FIG. 18B.

Initially, two threshold values of 128 kbps and 147 kbps are provided.It is considered by experience that if bit quantity (rate) is about 19kbps, such bit quantity is sufficient for sub information. Thus, 147kbps is set as the lowest limit by taking this bit quantity and bitquantity for data into consideration.

Moreover, in the case where bit allocation quantity required for acertain channel is above 128 kbps and is below 147 kbps, bits which canbe used for sub information are reduced by data portion above 128 kbps.In such case, with respect to this channel, there is carried out bitallocation which includes no subsidiary allocation bits mentioned above,and is smaller than 128 kbps and is as close as 128 kbps. By thisprocessing, sound quality is lowered to some extent. However, 19 kbps atmaximum, i.e., bit quantity corresponding to sub information exist asbit reduction quantity. When compatibility is taken into consideration,this method has greater merit.

The case where bit allocation quantity required for a certain channel isabove 147 kbps will now be described with reference to FIG. 18B.

Let consider that with respect to, e.g., channel CH1, bit quantity isdivided into two portions of the portion (1a) in which a certain fixedbit quantity, e.g., 128 kbps is maximum and portion (1b, S) above 128kbps.

Namely, it is now assumed that input signal is divided into the portionquantized by 128 kbps corresponding to independent allocation and theportion quantized by allocation beyond 128 kbps, and respective portionsare quantized by allocated bit quantities.

Similarly to the example of FIG. 18A, in FIG. 19, first quantizer 901and second quantizer 903 carry out bit allocation including round-offprocessing.

By two quantizers, higher order bits are quantized by bit allocationbelow 128 kbps and closer to 128 kbps, and are encoded.

On the other hand, lower order bits are quantized by bit allocation ofthe portion above 128 kbps, and are encoded.

Attention must be paid to the fact that there is limitation in bitquantity which can be used for subsidiary allocation in both cases ofFIGS. 18A and 18B.

For example, in the example of FIG. 18A, total bit quantity necessaryfor subsidiary allocation is sum of slanting line portions (S) of CH1,CH3 and CH6. In addition, sub information of respective channels must betaken into consideration.

On the contrary, if bit rates of all channels are assumed to be fixed,sum of excess (surplus) bits (R) of respective channels of CH2, CH4,CH5, CH7 and CH8 corresponds to maximum bit quantity which can be usedfor subsidiary allocation. If upper limit of bit rate is 800 kbps as inthis embodiment, bit quantity which can be used for subsidiaryallocation becomes considerably lesser.

Thus, in the case where excess bits are insufficient, limitation must begiven to bits which can be used for subsidiary allocation in dependencyupon priority rank. In a manner of the previously described example ofFIG. 22, allocation bit quantity is reduced (S15). In addition,employment of a method of uniformly reducing allocation quantity or amethod of carrying out allocation in the state where specific channel isconsidered as a preferential channel is also effective.

It is to be noted that, as previously described, with respect to scalefactors of data by the subsidiary allocation, since such scale factorscan be calculated from scale factors and word lengths of data by bitallocation corresponding to independent allocation, it is sufficient totransmit only word length.

Data of respective channels obtained at quantizers 901 and 903 in amanner as described above are arranged at sync block having apredetermined time as unit. Way of arrangement of data of respectivechannels is shown in a model form in FIG. 20.

In FIG. 20, data arrangement of the previously described example of FIG.18 is shown. Within sync block, initially (1) channel data which do notuse the subsidiary allocation, i.e., channel data (CH2, CH4, CH5, CH7,CH8) by bit allocation less than 128 kbps, which are indicated bylattice pattern, (2) channel data (CH1, CH3, CH6) of the portion inwhich a predetermined bit quantity, e.g., 128 Kbsp is maximum, which areshown by void, of channel data using the subsidiary allocation, and (3)channel data (CH1, CH3, CH6) of the portion by subsidiary allocationabove 128 kbps, which are indicated by slanting line pattern, of channeldata using the subsidiary allocation are arranged in this way, therebymaking it possible to carry out a processing as described below.

In decoder using no subsidiary allocation, only channel data of the (1)portion, and the (2) portion are used. Thus, with respect to allchannels, they can be dealt similarly to data using no subsidiaryallocation. Since data by subsidiary allocation is not used, channeldata (CH1, CH3, CH6) by subsidiary allocation would be, e.g., decodedata of only MSB portion. Thus, there results the state wherere-quantization has been carried out at coarse quantization step. As aresult, sound quality is deteriorated. However, such a deterioration isconsidered to be deterioration to such a degree that it does not becomeproblem from a viewpoint of the hearing sense.

In decoder using subsidiary allocation, all data are used. Thus, channeldata (CH1, CH3, CH6) by subsidiary allocation constitute complete wordin which MSB portion and LSB portion are synthesized. Accordingly, it ispossible to decode extremely high quality speech signals.

In this example, the case where the number of bits using subsidiaryallocation bits is 3 is illustrated. In practice, if determination ismade such that subsidiary allocation bits are used for only forward twochannels important from a viewpoint of sound quality, processing of syncblock is simplified. In contrast, in the case where channels usingsubsidiary allocation bits are not determined, i.e., channels usingsubsidiary allocation bits are adaptively switched in dependency uponinput signal, channel ID is added to respective data, thereby making itpossible to cope with such situation. For ID, there only resultsincrease in data of 3 bits per channel.

In addition, in FIG. 19, in decoder corresponding to encoder, gaincontroller 907 is provided in correspondence with gain controller 906,and gain controller 908 is provided in correspondence with gaincontroller 905. Outputs of these gain controllers 907, 908 are added atadder 904. Added output thus obtained is taken out from output terminal910.

Added output is added output of data by independent allocation and databy subsidiary allocation, and is caused to be complete data.

FIG. 21 shows a fundamental decoding apparatus of the embodiment of thisinvention for decoding efficiently encoded signals for a second time.

In FIG. 21, quantized MDCT coefficients of respective bands are given todecoding apparatus input terminals 122˜124, and used block sizeinformation are given to input terminals 125˜127. Decoding circuits116˜118 release bit allocation by using adaptive bit allocationinformation.

Then, signals in the frequency region are transformed into signals inthe time region at I-MDCT circuits 113˜115. These time region signals ofthe partial bands are decoded into entire band signals by I-QMF circuits112, 111.

In this example, respective portions of the portion in which 128 kbps iscaused to be minimum and the portion by subsidiary allocation bits inchannels where bit allocation (1) less than 128 kbps has been carriedout and channels where bit allocation (2) above 147 kbps has beencarried out are decoded at the decoding circuits 116˜118.

It should be noted that bit portions using subsidiary allocation arerespectively decoded, and are then caused to be one word as LSB portionand MSB portion, resulting in single high precision sample.

Recording media of the embodiment according to this invention areadapted so that signals encoded by efficient encoding apparatus of theembodiment of this invention as described above are recorded thereontoor thereinto. In addition to the previously described cinema film, therecan be enumerated recording media where the encoded signals are recordedon disc-shaped recording medium such as optical disc, magneto-opticaldisc or magnetic disc, etc., recording media where the encoded signalsare recorded on magnetic tape, etc., semiconductor memory or IC card inwhich encoded signals are stored, and the like.

Moreover, efficiently encoded signal recording method of the embodimentof this invention onto or into recording media of the embodiment of thisinvention records, in a separate manner, within one sync block, samplegroup relating to first bit allocation quantity to allocate bit quantitygreater than a fixed reference quantity for plural channels and theremaining second bit allocation sample group of the sample grouprelating to the first bit allocation quantity for plural channels.Further, such a recording is carried out alternately every respectivechannels.

In this invention, compression encoding/decoding apparatus for digitalspeech signal and method therefor have been described in detail as theembodiments.

In addition, while no explanation is given in detail as the embodiment,it is of course that this invention can be applied not only to digitalspeech signals, but also to digital picture signals.

Namely, in such systems to transmit/record a plurality of movingpictures by parallel channels, with respect to picture channels ofcomplicated pattern in which required bit allocation is great, it ispossible to similarly handle the portion above a predetermined value assubsidiary allocation.

Industrial Applicability

As apparent from the foregoing description, in the efficient encodingmethod according to this invention, the efficient code decoding methodcorresponding thereto, the efficient code decoding/reproducing method,efficiently encoded signal recording method for recording signalsencoded by that efficient encoding method, and recording media whererecording has been implemented, it is possible to reproduce, in thestate of high sound quality, by making use of dependent allocation,compressed signals caused to have improved sound quality by usingdependent allocation technology with respect to compression ofmulti-channel system.

Moreover, also in ordinarily frequently used decoder adapted forcarrying out bit allocation every channel by using bit rate less thanfixed value individually with respect to respective channel,reproduction can be made without great deterioration of sound quality.Further, from facts as above, speech signals on, e.g., cinema film canbe easily diverted to other optical disc media at the same time. Inaddition, signals on film can be prepared also by cheap and ordinarilyfrequently used decoders adapted for carrying out bit allocation everychannels by using bit rate less than fixed value individually withrespect to respective channels.

What is claimed is:
 1. An efficient encoding method of re-quantizingsample data of respective digital signals of a plurality of channels bya predetermined bit quantity to encode them,the method comprising thesteps of:specifying a channel in which bit quantity above a fixedreference quantity determined in advance is allocated; allocating a bitquantity allocated to the channel to a first bit quantity which is notabove the fixed reference quantity at most and the remaining second bitquantity; re-quantizing a portion of the sample data by using bits bythe first bit quantity; re-quantizing at least the other portion of thesample data by using bits by the second bit quantity; and synthesizingthe re-quantized one portion of the sample data and re-quantized theother portion of the sample data.
 2. An efficient encoding method as setforth in claim 1, wherein sum total at all channels of bit quantities ofbits allocated to the respective channels is substantially fixed.
 3. Thestep of requantizing the sample data of the respective channels of theefficient encoding method as set forth in claim 1, further including thesteps of:normalizing a plurality of sample data by a common scalefactor; and limiting word lengths of respective sample data normalizedby the common scale factor.
 4. An efficient encoding method as set forthin claim 3,wherein data by the first bit quantity and data by the secondbit quantity are positioned in different areas within sync block.
 5. Anefficient encoding method as set forth in claim 1,wherein the second bitquantity is bit allocation in which no subsidiary allocation bit isincluded, and the second bit allocation quantity is a difference betweenbit allocation in which subsidiary allocation bits are included and abit allocation in which the first bit allocation quantity is notincluded.
 6. An efficient encoding method as set forth in claim 1,wherein the second bit quantity is less than sum total bit quantity ofexcess bits.
 7. An efficient encoding method as set forth in claim 1,wherein the step of carrying out re-quantization re-quantizes sampledata every small blocks obtained by subdividing an input signal withrespect to a time base and a frequency base.
 8. An efficient encodingmethod as set forth in claim 7,wherein sample data within the smallblocks subdivided with respect to the time base and the frequency baseis caused to undergo analysis of non-blocking frequency characteristicto allow an output of the analysis of the non-blocking frequencycharacteristic to undergo analysis of blocking frequency characteristic.9. An efficient encoding method as set forth in claim 8, wherein, in theanalysis of the blocking frequency characteristic, block size isadaptively altered in dependency upon time characteristic of the inputsignal.
 10. An efficient encoding method as set forth in claim 1,wherein the fixed reference quantity determined in advance of the stepof specifying a channel is at least two reference quantities of areference quantity of bit quantity used for data and a referencequantity in which a bit quantity for sub information is taken intoconsideration.
 11. An efficient encoding method as set forth in claim 1,wherein the fixed reference quantity determined in advance of the stepof specifying a channel is a reference quantity in which a bit quantityfor the sub information is taken into consideration.
 12. An efficientencoding method as set forth in claim 1, wherein the allocation of bitsis based on independent allocation in which bits are independentlyallocated every respective channels and dependent allocation in whichbits are correlatively allocated between the respective channels.
 13. Anefficient encoding method as set forth in claim 12,wherein the first bitquantity is based on a bit allocation by the independent allocation, andwherein the second bit quantity is based on a bit allocation by thedependent allocation.
 14. An efficient encoding method as set forth inclaim 13, wherein alterations of block size are independently carriedout every output bands of analysis of at least two non-blockingfrequency characteristics.
 15. An efficient encoding method as set forthin claim 1, wherein sum of the first bit allocation portion and thesecond bit allocation portion of the respective channels changes bymaximum value of scale factors or sample data of the respectivechannels.
 16. An efficient encoding method as set forth in claim 12,wherein the dependent allocation is changed by changes in point of timeof amplitude information of energy values, peak values or mean values ofsignals of the respective channels.
 17. An efficient encoding method asset forth in claim 12, wherein the dependent allocation is changed bychanges in point of time of scale factors of the respective channels.18. A recording medium adapted so that encoded signals formed by theefficient encoding method as set forth in any one of claims 1 to 17 arerecorded thereon or therein.
 19. An efficiently encoded signal recordingmethod for recording data of a plurality of channels into a single syncblock,the method comprising the steps:recording data of a channel inwhich a bit quantity smaller than a fixed reference quantity determinedin advance is allocated; and recording data of a channel in which a bitquantity greater than a fixed reference quantity determined in advanceis allocated, the step of recording data further comprising the stepsof: allocating a bit quantity allocated to the channel to a first bitquantity which is not above the fixed reference quantity at most and theremaining second quantity; re-quantizing a portion of the sample data byusing bits by the first bit quantity to record the re-quantized data;and re-quantizing at least the other portion of the sample data by usingbits by the second bit quantity to record the re-quantized data.
 20. Anefficiently encoded signal recording method as set forth in claim 19,wherein the re-quantized data are alternately recorded within the syncblock.
 21. A decoding method for an efficiently encoded signal by anencoding method of re-quantizing sample data of respective digitalsignals of a plurality of channels by a predetermined bit quantity toencode them,the efficient encoding method comprising the stepsof:specifying a channel in which a bit quantity greater than a fixedreference quantity determined in advance is allocated; allocating a bitquantity allocated to the channel to a first bit quantity which is notabove the fixed reference quantity at most and the remaining second bitquantity; re-quantizing a portion of the sample data by using bits bythe first bit quantity; re-quantizing at least the other portion of thesample data by using bits by the second bit quantity; and synthesizing aportion of the re-quantized sample data and the other portion of there-quantized sample data, the synthesized sample data being recorded ortransmitted, sample data at least by the first bit quantity of therecorded or transmitted sample data being caused to be sample data ofthe corresponding channel to obtain a digital signal of at least one ofthe plurality of channels.
 22. A decoding method for an efficientlyencoded signal by an encoding method of re-quantizing sample data ofrespective digital signals of a plurality of channels by a predeterminedbit quantity to encode them,the encoding method comprising the stepsof:specifying a channel in which a bit quantity greater than a fixedreference quantity determined in advance is allocated; allocating a bitquantity allocated to the channel to a first bit quantity which is notabove the fixed reference quantity at most and the remaining second bitquantity; re-quantizing a portion of the sample data by using bits bythe first bit quantity; re-quantizing at least the other portion of thesample data by using bits by the second bit quantity; synthesizing aportion of the re-quantized sample data and the other portion of there-quantized sample data, the synthesized sample data being recorded ortransmitted, only sample data by the first bit quantity of the recordedor transmitted sample data being caused to be sample data of thecorresponding channel to obtain a digital signal of at least one of theplurality channels.
 23. The step of re-quantizing the sample data of therespective channels of the decoding method for efficiently encodedsignal as set forth in claims 21 and 22,further including the stepsof:normalizing a plurality of sample data by a common scale factor; andlimiting word lengths of the respective sample data normalized by thecommon scale factor.
 24. A decoding method for efficiently encodedsignal as set forth in claim 23, wherein scale factor for sample datarelating to the second bit allocation quantity is determined from scalefactors and word lengths for sample data relating to the first bitallocation quantity.
 25. An efficient encoding apparatus adapted forre-quantizing sample data of respective digital signals of a pluralityof channels by a predetermined bit quantity to encode them,the efficientencoding apparatus comprising:means for specifying a channel in which abit quantity greater than a fixed reference quantity determined inadvance is allocated; means for allocating a bit quantity allocated tothe channel to a first bit quantity which is not above the fixedreference quantity at most and the remaining second bit quantity; meansfor re-quantizing a portion of the sample data by using bits by thefirst bit quantity; means for re-quantizing at least the other portionof the sample data by using bits by the second bit quantity; and meansfor synthesizing a portion of the re-quantized sample data and the otherportion of the re-quantized sample data.
 26. An efficient encodingapparatus as set forth in claim 25, wherein sum total at all channels ofbit quantities of bits allocated to the respective channels issubstantially fixed.
 27. Means for re-quantizing the sample data of therespective channels of the efficient encoding apparatus as set forth inclaim 25 further including:means for normalizing a plurality of sampledata by a common scale factor; and means for limiting word lengths ofrespective sample data normalized by the common scale factor.
 28. Anefficient encoding apparatus as set forth in claim 25, wherein data bythe first bit quantity and data by the second bit quantity arepositioned in different areas within sync block.
 29. An efficientencoding apparatus as set forth in claim 25, wherein the second bitquantity is bit allocation in which no subsidiary bit is included, andthe second bit allocation quantity is a difference between bitallocation in which subsidiary allocation bits are included and bitallocation in which the first bit allocation quantity is not included.30. An efficient encoding apparatus as set forth in claim 25, whereinthe second bit quantity is less than sum total bit quantity of excessbits.
 31. An efficient encoding apparatus as set forth in claim 25,wherein the means for carrying out re-quantization re-quantizes sampledata every small blocks subdivided with respect to the time base and thefrequency base.
 32. An efficient encoding apparatus as set forth inclaim 31,wherein sample data within the small blocks subdivided withrespect to the time base and the frequency base is caused to undergoanalysis of non-blocking frequency characteristic to allow an output ofanalysis of the non-blocking frequency characteristic to undergoanalysis of blocking frequency characteristic.
 33. An efficient encodingapparatus as set forth in claim 32, wherein, in the analysis of theblocking frequency characteristic, block size is adaptively altered independency upon time characteristic of an input signal.
 34. An efficientencoding apparatus as set forth in claim 25, wherein the fixed referencequantity determined in advance is at least two reference quantities of areference quantity of bit quantity used for data and a referencequantity in which a bit quantity for sub information is taken intoconsideration.
 35. An efficient encoding apparatus as set forth in claim25, wherein the fixed reference quantity determined in advance is areference quantity in which a bit quantity for sub information is takeninto consideration, the apparatus including means for specifying achannel in which a bit quantity greater than the reference quantity isallocated.
 36. An efficient encoding apparatus as set forth in claim 25,wherein the allocation of bits is based on independent allocation inwhich bits are independently allocated every respective channels anddependent allocation in which bits are correlatively allocated betweenthe respective channels.
 37. An efficient encoding apparatus as setforth in claim 36,wherein the first bit quantity is based on bitallocation by the independent allocation, and wherein the second bitquantity is based on bit allocation by the dependent allocation.
 38. Anefficient encoding apparatus as set forth in claim 35, whereinalteration of block size is independently carried out every output bandsof analysis of at least two non-blocking frequency characteristics. 39.An efficient encoding apparatus as set forth in claim 25, wherein sum ofthe first bit allocation portion and the second bit allocation portionof the respective channels change by maximum value of scale factors orsample data of the respective channels.
 40. An efficient encodingapparatus as set forth in claim 36, wherein dependent allocation ischanged by changes in point of time of amplitude information of energyvalues, peak values or mean values of signals of the respectivechannels.
 41. An efficient encoding apparatus as set forth in claim 36,wherein dependent allocation is changed by changes in point of time ofscale factors of the respective channels.
 42. A recording medium adaptedso that encoded signals formed by the efficient encoding apparatus asset forth in any one of claims 25 to 39 are recorded thereonto ortherein.
 43. A recording medium adapted so that encoded signals as setforth in claim 42 are recorded thereon or therein, wherein the recordingmedium is cinema film.
 44. An efficient encoding/decoding system forre-quantizing sample data of respective digital signals of a pluralityof channels by a predetermined bit quantity to encode re-quantized datato decode encoded data,the efficient encoding/decoding systemcomprising:means for specifying a channel in which a bit quantitygreater than a fixed reference quantity determined in advance isallocated; means for allocating a bit quantity allocated to the channelto a first bit quantity which is not above the fixed reference quantityat most and the remaining second bit quantity; means for re-quantizing aportion of the sample data by using bits by the first bit quantity;means for re-quantizing at least the other portion of the sample data byusing bits by the second bit quantity; means for synthesizing a portionof the re-quantized sample data and the other portion of there-quantized sample data; means for recording the synthesized sampledata onto or into a recording medium or transmitting it to atransmission path; and means for decoding sample data obtained byre-quantizing a portion of sample data by using the sample data by atleast the first bit quantity of the recorded or transmitted sample datato obtain a digital of at least one of the plurality of channels.
 45. Anefficient encoding/decoding system for re-quantizing sample data ofrespective digital signals of a plurality of channels by a predeterminedbit quantity to encode re-quantized data to decode encoded data,theefficient encoding/decoding system comprising:means for specifying achannel in which a bit quantity greater than a fixed reference quantitydetermined in advance is allocated; means for allocating a bit quantityallocated to the channel to a first bit quantity which is not above thefixed reference quantity at most and the remaining second bit quantity;means for re-quantizing a portion of the sample data by using bits bythe first bit quantity; means for re-quantizing at least the otherportion of the sample data by using bits by the second bit quantity;means for synthesizing a portion of the re-quantized sample data and theother portion of the re-quantized sample data; means for recording thesynthesized sample data onto or into a recording medium, or transmittingit to a transmission path; and means for decoding, as sample data of thecorresponding channel, only sample data by the first bit quantity of therecorded or transmitted sample data to obtain a digital signal of atleast one of the plurality of channels.